Remove AudioCodingModule::Process()
An earlier change moved the encoding work from Process to
Add10MsData; process was just a no-op.
BUG=3520
COAUTHOR=kwiberg@webrtc.org
R=henrika@webrtc.org, minyue@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43439004
Cr-Commit-Position: refs/heads/master@{#8553}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8553 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
index c1b1636..e9096ff 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
@@ -96,17 +96,13 @@
frame.num_channels_ = codec.channels;
memset(frame.data_, 0, frame.samples_per_channel_ * frame.num_channels_ *
sizeof(int16_t));
- int num_bytes = 0;
packet_sent_ = false;
last_packet_send_timestamp_ = timestamp_;
- while (num_bytes == 0) {
+ while (!packet_sent_) {
frame.timestamp_ = timestamp_;
timestamp_ += frame.samples_per_channel_;
- ASSERT_EQ(0, acm_->Add10MsData(frame));
- num_bytes = acm_->Process();
- ASSERT_GE(num_bytes, 0);
+ ASSERT_GE(acm_->Add10MsData(frame), 0);
}
- ASSERT_TRUE(packet_sent_); // Sanity check.
}
// Last element of id should be negative.
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc
index 6749e12..41e0feb 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc
@@ -80,10 +80,10 @@
input_frame_.num_channels_,
input_frame_.data_);
}
- CHECK_EQ(0, acm_->Add10MsData(input_frame_));
+ data_to_send_ = false;
+ CHECK_GE(acm_->Add10MsData(input_frame_), 0);
input_frame_.timestamp_ += input_block_size_samples_;
- int32_t encoded_bytes = acm_->Process();
- if (encoded_bytes > 0) {
+ if (data_to_send_) {
// Encoded packet received.
return CreatePacket();
}
@@ -106,6 +106,7 @@
timestamp_ = timestamp;
last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes);
assert(last_payload_vec_.size() == payload_len_bytes);
+ data_to_send_ = true;
return 0;
}
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h
index 80aaf36..05a29df 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h
@@ -79,6 +79,7 @@
uint32_t timestamp_;
uint16_t sequence_number_;
std::vector<uint8_t> last_payload_vec_;
+ bool data_to_send_;
DISALLOW_COPY_AND_ASSIGN(AcmSendTestOldApi);
};
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index 542aaba..62287a4 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -145,7 +145,6 @@
aux_rtp_header_(NULL),
receiver_initialized_(false),
first_10ms_data_(false),
- last_encode_value_(0),
callback_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
packetization_callback_(NULL),
vad_callback_(NULL) {
@@ -218,11 +217,6 @@
"Destroyed");
}
-int32_t AudioCodingModuleImpl::Process() {
- CriticalSectionScoped lock(acm_crit_sect_);
- return last_encode_value_;
-}
-
int32_t AudioCodingModuleImpl::Encode() {
// Make room for 1 RED payload.
uint8_t stream[2 * MAX_PAYLOAD_SIZE_BYTE];
@@ -752,15 +746,7 @@
// Add 10MS of raw (PCM) audio data to the encoder.
int AudioCodingModuleImpl::Add10MsData(const AudioFrame& audio_frame) {
int r = Add10MsDataInternal(audio_frame);
- if (r < 0) {
- CriticalSectionScoped lock(acm_crit_sect_);
- last_encode_value_ = -1;
- } else {
- int r_encode = Encode();
- CriticalSectionScoped lock(acm_crit_sect_);
- last_encode_value_ = r_encode;
- }
- return r;
+ return r < 0 ? r : Encode();
}
int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame) {
@@ -1563,10 +1549,9 @@
}
int AudioCodingImpl::Add10MsAudio(const AudioFrame& audio_frame) {
- if (acm_old_->Add10MsData(audio_frame) != 0) {
+ if (acm_old_->Add10MsDataInternal(audio_frame) != 0)
return -1;
- }
- return acm_old_->Process();
+ return acm_old_->Encode();
}
const ReceiverInfo* AudioCodingImpl::GetReceiverInfo() const {
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
index 3450194..76d1e36 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -24,6 +24,7 @@
namespace webrtc {
class CriticalSectionWrapper;
+class AudioCodingImpl;
namespace acm2 {
@@ -32,12 +33,11 @@
class AudioCodingModuleImpl : public AudioCodingModule {
public:
+ friend webrtc::AudioCodingImpl;
+
explicit AudioCodingModuleImpl(const AudioCodingModule::Config& config);
~AudioCodingModuleImpl();
- // Process any pending tasks such as timeouts.
- virtual int32_t Process() OVERRIDE;
-
/////////////////////////////////////////
// Sender
//
@@ -335,7 +335,6 @@
AudioFrame preprocess_frame_ GUARDED_BY(acm_crit_sect_);
bool first_10ms_data_ GUARDED_BY(acm_crit_sect_);
- int last_encode_value_ GUARDED_BY(acm_crit_sect_);
CriticalSectionWrapper* callback_crit_sect_;
AudioPacketizationCallback* packetization_callback_
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
index a1ea179..122d027 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
@@ -193,15 +193,15 @@
}
virtual void InsertAudio() {
- ASSERT_EQ(0, acm_->Add10MsData(input_frame_));
+ ASSERT_GE(acm_->Add10MsData(input_frame_), 0);
+ VerifyEncoding();
input_frame_.timestamp_ += kNumSamples10ms;
}
- virtual void Encode() {
- int32_t encoded_bytes = acm_->Process();
- // Expect to get one packet with two bytes per sample, or no packet at all,
- // depending on how many 10 ms blocks go into |codec_.pacsize|.
- EXPECT_TRUE(encoded_bytes == 2 * codec_.pacsize || encoded_bytes == 0);
+ virtual void VerifyEncoding() {
+ int last_length = packet_cb_.last_payload_len_bytes();
+ EXPECT_TRUE(last_length == 2 * codec_.pacsize || last_length == 0)
+ << "Last encoded packet was " << last_length << " bytes.";
}
const int id_;
@@ -316,7 +316,6 @@
if (packet_cb_.num_calls() > 0)
EXPECT_EQ(kAudioFrameSpeech, packet_cb_.last_frame_type());
InsertAudio();
- Encode();
}
EXPECT_EQ(kLoops / k10MsBlocksPerPacket, packet_cb_.num_calls());
EXPECT_EQ(kAudioFrameSpeech, packet_cb_.last_frame_type());
@@ -330,13 +329,6 @@
class AudioCodingModuleTestWithComfortNoiseOldApi
: public AudioCodingModuleTestOldApi {
protected:
- void Encode() override {
- int32_t encoded_bytes = acm_->Process();
- // Expect either one packet with 9 comfort noise parameters, or no packet
- // at all.
- EXPECT_TRUE(encoded_bytes == 9 || encoded_bytes == 0);
- }
-
void RegisterCngCodec(int rtp_payload_type) {
CodecInst codec;
AudioCodingModule::Codec("CN", &codec, kSampleRateHz, 1);
@@ -345,6 +337,12 @@
ASSERT_EQ(0, acm_->RegisterSendCodec(codec));
}
+ void VerifyEncoding() override {
+ int last_length = packet_cb_.last_payload_len_bytes();
+ EXPECT_TRUE(last_length == 9 || last_length == 0)
+ << "Last encoded packet was " << last_length << " bytes.";
+ }
+
void DoTest(int blocks_per_packet, int cng_pt) {
const int kLoops = 40;
// This array defines the expected frame types, and when they should arrive.
@@ -371,7 +369,6 @@
int num_calls_before = packet_cb_.num_calls();
EXPECT_EQ(i / blocks_per_packet, num_calls_before);
InsertAudio();
- Encode();
int num_calls = packet_cb_.num_calls();
if (num_calls == num_calls_before + 1) {
EXPECT_EQ(expectation[num_calls - 1].ix, i);
@@ -493,7 +490,6 @@
}
++send_count_;
InsertAudio();
- Encode();
if (TestDone()) {
test_complete_->Set();
}
@@ -586,7 +582,6 @@
int loop_counter = 0;
while (packet_cb_.last_payload_len_bytes() == 0) {
InsertAudio();
- Encode();
ASSERT_LT(loop_counter++, 10);
}
// Set |last_packet_number_| to one less that |num_calls| so that the packet
@@ -632,7 +627,9 @@
AudioCodingModuleTestOldApi::InsertAudio();
}
- void Encode() { ASSERT_GE(acm_->Process(), 0); }
+ // Override the verification function with no-op, since iSAC produces variable
+ // payload sizes.
+ void VerifyEncoding() override {}
// This method is the same as AudioCodingModuleMtTestOldApi::TestDone(), but
// here it is using the constants defined in this class (i.e., shorter test
diff --git a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
index f14ff80..72e00cd 100644
--- a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
@@ -101,8 +101,6 @@
static AudioCodingModule* Create(int id, Clock* clock);
virtual ~AudioCodingModule() {};
- virtual int32_t Process() = 0;
-
///////////////////////////////////////////////////////////////////////////
// Utility functions
//
@@ -317,9 +315,12 @@
///////////////////////////////////////////////////////////////////////////
// int32_t Add10MsData()
- // Add 10MS of raw (PCM) audio data to the encoder. If the sampling
+ // Add 10MS of raw (PCM) audio data and encode it. If the sampling
// frequency of the audio does not match the sampling frequency of the
- // current encoder ACM will resample the audio.
+ // current encoder ACM will resample the audio. If an encoded packet was
+ // produced, it will be delivered via the callback object registered using
+ // RegisterTransportCallback, and the return value from this function will
+ // be the number of bytes encoded.
//
// Input:
// -audio_frame : the input audio frame, containing raw audio
@@ -328,10 +329,8 @@
// AudioFrame.
//
// Return value:
- // 0 successfully added the frame.
- // -1 some error occurred and data is not added.
- // < -1 to add the frame to the buffer n samples had to be
- // overwritten, -n is the return value in this case.
+ // >= 0 number of bytes encoded.
+ // -1 some error occurred.
//
virtual int32_t Add10MsData(const AudioFrame& audio_frame) = 0;
diff --git a/webrtc/modules/audio_coding/main/test/APITest.cc b/webrtc/modules/audio_coding/main/test/APITest.cc
index 6893587..cf21343 100644
--- a/webrtc/modules/audio_coding/main/test/APITest.cc
+++ b/webrtc/modules/audio_coding/main/test/APITest.cc
@@ -411,33 +411,11 @@
bool APITest::ProcessRunA() {
_processEventA->Wait(100);
- if (_acmA->Process() < 0) {
- // do not print error message if there is no encoder
- bool thereIsEncoder;
- {
- ReadLockScoped rl(_apiTestRWLock);
- thereIsEncoder = _thereIsEncoderA;
- }
-
- if (thereIsEncoder) {
- fprintf(stderr, "\n>>>>> Process Failed at A <<<<<\n");
- }
- }
return true;
}
bool APITest::ProcessRunB() {
_processEventB->Wait(100);
- if (_acmB->Process() < 0) {
- bool thereIsEncoder;
- {
- ReadLockScoped rl(_apiTestRWLock);
- thereIsEncoder = _thereIsEncoderB;
- }
- if (thereIsEncoder) {
- fprintf(stderr, "\n>>>>> Process Failed at B <<<<<\n");
- }
- }
return true;
}
diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
index eff458b..b785cc3 100644
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
+++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
@@ -100,11 +100,8 @@
if (!_pcmFile.EndOfFile()) {
EXPECT_GT(_pcmFile.Read10MsData(_audioFrame), 0);
int32_t ok = _acm->Add10MsData(_audioFrame);
- EXPECT_EQ(0, ok);
- if (ok != 0) {
- return false;
- }
- return true;
+ EXPECT_GE(ok, 0);
+ return ok >= 0 ? true : false;
}
return false;
}
@@ -114,7 +111,6 @@
if (!Add10MsData()) {
break;
}
- EXPECT_GT(_acm->Process(), -1);
}
}
diff --git a/webrtc/modules/audio_coding/main/test/SpatialAudio.cc b/webrtc/modules/audio_coding/main/test/SpatialAudio.cc
index 7286fc1..b28c510 100644
--- a/webrtc/modules/audio_coding/main/test/SpatialAudio.cc
+++ b/webrtc/modules/audio_coding/main/test/SpatialAudio.cc
@@ -171,9 +171,6 @@
}
CHECK_ERROR(_acmRight->Add10MsData(audioFrame));
- CHECK_ERROR(_acmLeft->Process());
- CHECK_ERROR(_acmRight->Process());
-
CHECK_ERROR(_acmReceiver->PlayoutData10Ms(outFileSampFreq, &audioFrame));
_outFile.Write10MsData(audioFrame);
}
@@ -190,8 +187,6 @@
_inFile.Read10MsData(audioFrame);
CHECK_ERROR(_acmLeft->Add10MsData(audioFrame));
- CHECK_ERROR(_acmLeft->Process());
-
CHECK_ERROR(_acmReceiver->PlayoutData10Ms(outFileSampFreq, &audioFrame));
_outFile.Write10MsData(audioFrame);
}
diff --git a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc b/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
index 17186b36..b1badb6 100644
--- a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
+++ b/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
@@ -434,9 +434,6 @@
infile_a_.Read10MsData(audio_frame);
CHECK_ERROR(acm_a_->Add10MsData(audio_frame));
- // Run sender side of ACM.
- CHECK_ERROR(acm_a_->Process());
-
// Verify that the received packet size matches the settings.
receive_size = channel->payload_size();
if (receive_size) {
diff --git a/webrtc/modules/audio_coding/main/test/TestRedFec.cc b/webrtc/modules/audio_coding/main/test/TestRedFec.cc
index b69a1ed..8f6ef9d 100644
--- a/webrtc/modules/audio_coding/main/test/TestRedFec.cc
+++ b/webrtc/modules/audio_coding/main/test/TestRedFec.cc
@@ -311,8 +311,7 @@
while (!_inFileA.EndOfFile()) {
EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
- EXPECT_EQ(0, _acmA->Add10MsData(audioFrame));
- EXPECT_GT(_acmA->Process(), -1);
+ EXPECT_GE(_acmA->Add10MsData(audioFrame), 0);
EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame));
_outFileB.Write10MsData(audioFrame.data_, audioFrame.samples_per_channel_);
msecPassed += 10;
diff --git a/webrtc/modules/audio_coding/main/test/TestStereo.cc b/webrtc/modules/audio_coding/main/test/TestStereo.cc
index 5e7bc51..72ae25e 100644
--- a/webrtc/modules/audio_coding/main/test/TestStereo.cc
+++ b/webrtc/modules/audio_coding/main/test/TestStereo.cc
@@ -779,10 +779,7 @@
}
in_file_stereo_->Read10MsData(audio_frame);
}
- EXPECT_EQ(0, acm_a_->Add10MsData(audio_frame));
-
- // Run sender side of ACM
- EXPECT_GT(acm_a_->Process(), -1);
+ EXPECT_GE(acm_a_->Add10MsData(audio_frame), 0);
// Verify that the received packet size matches the settings.
rec_size = channel->payload_size();
diff --git a/webrtc/modules/audio_coding/main/test/TestVADDTX.cc b/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
index 1e05be2..20ccde6 100644
--- a/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
+++ b/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
@@ -242,8 +242,7 @@
_inFileA.Read10MsData(audioFrame);
audioFrame.timestamp_ = timestampA;
timestampA += SamplesIn10MsecA;
- EXPECT_EQ(0, _acmA->Add10MsData(audioFrame));
- EXPECT_GT(_acmA->Process(), -1);
+ EXPECT_GE(_acmA->Add10MsData(audioFrame), 0);
EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame));
_outFileB.Write10MsData(audioFrame.data_, audioFrame.samples_per_channel_);
}
diff --git a/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc b/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc
index cc0fc20..f4db1db 100644
--- a/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc
+++ b/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc
@@ -282,32 +282,21 @@
// and B, APIs will be called, like ResetEncoder(), and the code should
// continue to run, and be able to recover.
bool expect_error_add = false;
- bool expect_error_process = false;
while (!_inFileA.EndOfFile() && !_inFileB.EndOfFile()) {
msecPassed += 10;
EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
- EXPECT_EQ(0, _acmA->Add10MsData(audioFrame));
- EXPECT_EQ(0, _acmRefA->Add10MsData(audioFrame));
+ EXPECT_GE(_acmA->Add10MsData(audioFrame), 0);
+ EXPECT_GE(_acmRefA->Add10MsData(audioFrame), 0);
EXPECT_GT(_inFileB.Read10MsData(audioFrame), 0);
// Expect call to pass except for the time when no send codec is registered.
if (!expect_error_add) {
- EXPECT_EQ(0, _acmB->Add10MsData(audioFrame));
+ EXPECT_GE(_acmB->Add10MsData(audioFrame), 0);
} else {
EXPECT_EQ(-1, _acmB->Add10MsData(audioFrame));
}
- // Expect to pass except for the time when there either is no send codec
- // registered, or no receive codec.
- if (!expect_error_process) {
- EXPECT_GT(_acmB->Process(), -1);
- } else {
- EXPECT_EQ(_acmB->Process(), -1);
- }
- EXPECT_EQ(0, _acmRefB->Add10MsData(audioFrame));
- EXPECT_GT(_acmA->Process(), -1);
- EXPECT_GT(_acmRefA->Process(), -1);
- EXPECT_GT(_acmRefB->Process(), -1);
+ EXPECT_GE(_acmRefB->Add10MsData(audioFrame), 0);
EXPECT_EQ(0, _acmA->PlayoutData10Ms(outFreqHzA, &audioFrame));
_outFileA.Write10MsData(audioFrame);
EXPECT_EQ(0, _acmRefA->PlayoutData10Ms(outFreqHzA, &audioFrame));
@@ -328,14 +317,12 @@
EXPECT_EQ(0, _acmA->ResetEncoder());
EXPECT_EQ(0, _acmB->InitializeSender());
expect_error_add = true;
- expect_error_process = true;
}
// Re-register send codec on side B.
if (((secPassed % 5) == 4) && (msecPassed >= 990)) {
EXPECT_EQ(0, _acmB->RegisterSendCodec(codecInst_B));
EXPECT_EQ(0, _acmB->SendCodec(&dummy));
expect_error_add = false;
- expect_error_process = false;
}
// Reset decoder on side B, and initialize receiver on side A.
if (((secPassed % 7) == 6) && (msecPassed == 0)) {
diff --git a/webrtc/modules/audio_coding/main/test/delay_test.cc b/webrtc/modules/audio_coding/main/test/delay_test.cc
index 55dfd93..0f9e8b0 100644
--- a/webrtc/modules/audio_coding/main/test/delay_test.cc
+++ b/webrtc/modules/audio_coding/main/test/delay_test.cc
@@ -209,8 +209,7 @@
}
in_file_a_.Read10MsData(audio_frame);
- ASSERT_EQ(0, acm_a_->Add10MsData(audio_frame));
- ASSERT_LE(0, acm_a_->Process());
+ ASSERT_GE(acm_a_->Add10MsData(audio_frame), 0);
ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
out_file_b_.Write10MsData(
audio_frame.data_,
diff --git a/webrtc/modules/audio_coding/main/test/iSACTest.cc b/webrtc/modules/audio_coding/main/test/iSACTest.cc
index c5da92e..26236f2 100644
--- a/webrtc/modules/audio_coding/main/test/iSACTest.cc
+++ b/webrtc/modules/audio_coding/main/test/iSACTest.cc
@@ -246,10 +246,8 @@
void ISACTest::Run10ms() {
AudioFrame audioFrame;
EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
- EXPECT_EQ(0, _acmA->Add10MsData(audioFrame));
- EXPECT_EQ(0, _acmB->Add10MsData(audioFrame));
- EXPECT_GT(_acmA->Process(), -1);
- EXPECT_GT(_acmB->Process(), -1);
+ EXPECT_GE(_acmA->Add10MsData(audioFrame), 0);
+ EXPECT_GE(_acmB->Add10MsData(audioFrame), 0);
EXPECT_EQ(0, _acmA->PlayoutData10Ms(32000, &audioFrame));
_outFileA.Write10MsData(audioFrame);
EXPECT_EQ(0, _acmB->PlayoutData10Ms(32000, &audioFrame));
diff --git a/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc b/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
index de87b51..3d0d312 100644
--- a/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
+++ b/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
@@ -145,8 +145,7 @@
while (rms < kAmp / 2) {
in_audio_frame.timestamp_ = timestamp;
timestamp += in_audio_frame.samples_per_channel_;
- ASSERT_EQ(0, acm_a_->Add10MsData(in_audio_frame));
- ASSERT_LE(0, acm_a_->Process());
+ ASSERT_GE(acm_a_->Add10MsData(in_audio_frame), 0);
ASSERT_EQ(0, acm_b_->PlayoutData10Ms(codec.plfreq, &out_audio_frame));
rms = FrameRms(out_audio_frame);
++num_frames;
diff --git a/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc b/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
index 94b093d..7331696 100644
--- a/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
+++ b/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
@@ -164,7 +164,7 @@
// Push in just enough audio.
for (int n = 0; n < num_10ms_in_codec_frame_; n++) {
pcm_in_fid_.Read10MsData(frame_);
- EXPECT_EQ(0, send_acm_->Add10MsData(frame_));
+ EXPECT_GE(send_acm_->Add10MsData(frame_), 0);
}
// Set the parameters for the packet to be pushed in receiver ACM right
@@ -179,10 +179,6 @@
lost = true;
}
- // Process audio in send ACM, this should result in generation of a
- // packet.
- EXPECT_GT(send_acm_->Process(), 0);
-
if (FLAGS_verbose) {
if (!lost) {
std::cout << "\nInserting packet number " << seq_num
diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/modules/utility/source/coder.cc
index 5a39748..b106c75 100644
--- a/webrtc/modules/utility/source/coder.cc
+++ b/webrtc/modules/utility/source/coder.cc
@@ -96,10 +96,6 @@
return -1;
}
_encodedData = encodedData;
- if(_acm->Process() == -1)
- {
- return -1;
- }
encodedLengthInBytes = _encodedLengthInBytes;
return 0;
}
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 927440a..0a5d8e6 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -3560,7 +3560,10 @@
// The ACM resamples internally.
_audioFrame.timestamp_ = _timeStamp;
- if (audio_coding_->Add10MsData((AudioFrame&)_audioFrame) != 0)
+ // This call will trigger AudioPacketizationCallback::SendData if encoding
+ // is done and payload is ready for packetization and transmission.
+ // Otherwise, it will return without invoking the callback.
+ if (audio_coding_->Add10MsData((AudioFrame&)_audioFrame) < 0)
{
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::EncodeAndSend() ACM encoding failed");
@@ -3568,12 +3571,7 @@
}
_timeStamp += _audioFrame.samples_per_channel_;
-
- // --- Encode if complete frame is ready
-
- // This call will trigger AudioPacketizationCallback::SendData if encoding
- // is done and payload is ready for packetization and transmission.
- return audio_coding_->Process();
+ return 0;
}
int Channel::RegisterExternalMediaProcessing(
